Re: [sipx-users] auto-attendant messages are played real fast

I think the reINVITe came from sipXbridge. Not so sure if this is something hard-coded but perhaps it's the reason why ptime is set to 20 in freeswitch config as the default in the first place. Bottom line, this is not really the fault of either sipx or freeswitch. Your provider is not comp

Re: [sipx-users] SBC for port translation

Just thought I'd throw this out there... We have a Patton 4960 gateway that has a public IP address (WAN port) that uses IP based registration to connect to voip.ms (in the routing table - we only allow traffic on the WAN port to/from the ip address of the voip.ms server we are registering to).

Re: [sipx-users] auto-attendant messages are played real fast

Problem is still there even after changing the "codec-ms" value to 30.. 1) I changed the "param name="codec-ms" value="30"/".And rebooted the system. System prompted me to run "freeswitch.sh --configtest". And the issue (autoattendant messages are played at very fast rate) there. 2) Afterth

Re: [sipx-users] grandstream gxc2000 firmware?

On 1/27/11 9:14 PM, Michael Scheidell wrote: sipx->diagnosics,-> show registrations doesn't show it either. so, its NOT a 4 line phone! no, found the trick. if I look at the phone on its GUI, it wants a different local port per line. line 1, 5060 line 2, 5062, line 3, 5064, line 4, 5066 I wen

Re: [sipx-users] grandstream gxc2000 firmware?

On 1/27/11 6:30 PM, Michael Scheidell wrote: if a person was really into self abuse, and didn't learn their lessons from trying to get the cisco's to work, and wanted to install a couple of granstream, 4 link, gxc2000's... what firmware would you recommend for sipx 4.2.0? looks like it has al

Re: [sipx-users] HA - remote workers - public port

You would need to static NAT both servers to their own IP addresses and this should work... Mike On Wed, Jan 26, 2011 at 9:08 PM, Irena Dolovčak wrote: > @Tony: > > It is not that. I'm aware of 4.2.1 DNS slave misconfiguration (all times > expire in 1800seconds). I have manually changed the time

Re: [sipx-users] grandstream gxc2000 firmware?

As long as you don't expect BLF to work EVER then the Grandstreams work OK. The latest firmware available will do. From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] on behalf of Michael Scheidell [michael.scheid...@secnap.com]

[sipx-users] grandstream gxc2000 firmware?

if a person was really into self abuse, and didn't learn their lessons from trying to get the cisco's to work, and wanted to install a couple of granstream, 4 link, gxc2000's... what firmware would you recommend for sipx 4.2.0? looks like it has all the goodies: G.722, G.729(a/b), as well as

Re: [sipx-users] SBC for port translation

On 1/27/11 4:46 PM, Matt White wrote: I'm sure it offers no consolation, but we have several customers (including our personal corporate system) using VOIP.MS. We do not, and have never used any other translation than a plain port forward for inbound 5060 to translate to 5080. and you are usi

Re: [sipx-users] SBC for port translation

>>> Michael Scheidell 01/27/11 4:29 PM >>> >> >>I would be willing to try again, as EVERY NOW AND THEN, level3 complains about my 'experimental installation'. >>I want to put up a HA/DR vmail/call fwd system in a different geographic location, but am afraid the port iss

Re: [sipx-users] SBC for port translation

On 1/27/11 4:02 PM, Matt White wrote: Made this new thread. I'm interested in the scenario you point out here as I've never seen an second SBC required to translate 5060 to 5080 for sip trunking to work. Where do you see this scenario occur? I will check, but Tony and I worked hard on this (

Re: [sipx-users] SBC for port translation

On 1/27/2011 3:02 PM, Matt White wrote: >>> Tony Graziano >> When an invite comes from the provider and says the address and port "123.456.789.10:5080" and it needs to say instead >>"123.456.789.10:5060", its not the PACKET that needs to be translated or the address, its what is INSIDE the head

Re: [sipx-users] Polycom 3.2.4 transfer issue

Fs is not a proxy and cannot take the place of one. Even when using fs as an sbc, yet another package has to be added to do all the proxy/routing functions. Its (fs) onlly good is for its libraries. Been there, done that. Tony Graziano, Manager Telephone: 434.984.8430 F

[sipx-users] SBC for port translation

>>> Tony Graziano >> When an invite comes from the provider and says the address and port >> "123.456.789.10:5080" and it needs to say instead >>"123.456.789.10:5060", its not the PACKET that needs to be translated or the >>address, its what is INSIDE the header, not >>the actual packet destinat

Re: [sipx-users] Polycom 3.2.4 transfer issue

On 1/27/11 3:34 PM, Tony Graziano wrote: When an invite comes from the provider and says the address and port "123.456.789.10:5080" and it needs to say instead "123.456.789.10:5060", its not the PACKET that needs to be translated or the address, its what is INSIDE the header, not the actual pac

Re: [sipx-users] Polycom 3.2.4 transfer issue

On Thu, Jan 27, 2011 at 3:23 PM, Matthew Kitchin (public/usenet) < mkitchin.pub...@gmail.com> wrote: > On 1/27/2011 2:20 PM, Matt White wrote: > > > Do you have any traces with the firmware at 3.2.4? > > The traces you list have names of 3.2.1 and 3.2.4 but all the traces show > the phones still

Re: [sipx-users] Polycom 3.2.4 transfer issue

On 1/27/2011 2:20 PM, Matt White wrote: Do you have any traces with the firmware at 3.2.4? The traces you list have names of 3.2.1 and 3.2.4 but all the traces show the phones still running 3.2.1 Will do. I'm not sure how that could have happened. I say this because this issue looks much li

Re: [sipx-users] Polycom 3.2.4 transfer issue

Do you have any traces with the firmware at 3.2.4? The traces you list have names of 3.2.1 and 3.2.4 but all the traces show the phones still running 3.2.1 I say this because this issue looks much like this bug. http://track.sipfoundry.org/browse/XTRN-942 where the phone ignores the route and

Re: [sipx-users] Polycom 3.2.4 transfer issue

On 1/27/2011 1:22 PM, Matt White wrote: >>> "Matthew Kitchin (public/usenet)" only route, or does it NAT? I would suspect its actually NATing a 172.x.x.x IP to the 10.83 side if your changing the port. 01/27/11 2:02 PM >>> >>On 1/27/2011 12:53 PM, Matt White wrote: It is not NATing the IP.

Re: [sipx-users] Polycom 3.2.4 transfer issue

>>> "Matthew Kitchin (public/usenet)" only >>> route, or does it NAT? I would suspect its actually NATing a >>> 172.x.x.x IP to the 10.83 side if your changing the port. 01/27/11 2:02 PM >>> >>On 1/27/2011 12:53 PM, Matt White wrote:It is not NATing the

Re: [sipx-users] Polycom 3.2.4 transfer issue

On 1/27/11 2:04 PM, Matthew Kitchin (public/usenet) wrote: I cannot use voip.ms here. that was just a easy way to test it. even without remote workers, neither tony nor I could get voip.ms in ip authentication to work. (they INSIST on sending to port 5060 in ip authentication mode. no other o

Re: [sipx-users] Polycom 3.2.4 transfer issue

On 1/27/2011 12:57 PM, Michael Scheidell wrote: On 1/27/11 1:39 PM, Matthew Kitchin (public/usenet) wrote: I can shed some light on that I think. Verizon required they send us calls on port 5060. We do nat the port to 5080. We hadn't seen any issue there, but I guess we have now for some reas

Re: [sipx-users] Polycom 3.2.4 transfer issue

On 1/27/2011 12:53 PM, Matt White wrote: >>> "Matthew Kitchin (public/usenet)" 01/27/11 1:42 PM >>> >>There is no IP NAT. It is a private MPLS connection to Verizon. They use public IP ranges in these setups, but there is no >>IP nat involved. Their 172.x network and our 10.x network in this ex

Re: [sipx-users] Polycom 3.2.4 transfer issue

On 1/27/11 1:39 PM, Matthew Kitchin (public/usenet) wrote: I can shed some light on that I think. Verizon required they send us calls on port 5060. We do nat the port to 5080. We hadn't seen any issue there, but I guess we have now for some reason. Verizon --5060-- our router -- 5080 --> sipx

Re: [sipx-users] Polycom 3.2.4 transfer issue

On 1/27/2011 12:39 PM, Matthew Kitchin (public/usenet) wrote: On 1/27/2011 12:27 PM, Matt White wrote: >>> "Matthew Kitchin (public/usenet)" 01/27/11 12:34 PM >>> >>I think I see where the problem occurs in the sip trace, but I don't >>have the sip knowledge to understand why. >> >>Any help wo

Re: [sipx-users] Polycom 3.2.4 transfer issue

>>> "Matthew Kitchin (public/usenet)" 01/27/11 1:42 >>> PM >>> >>There is no IP NAT. It is a private MPLS connection to Verizon. Theyuse public IP ranges in these setups, but there is no >>IP nat involved. Their 172.x network and our 10.x network in this examplehave dire

Re: [sipx-users] Polycom 3.2.4 transfer issue

On 1/27/2011 12:37 PM, Matt White wrote: >>> Tony Graziano 01/27/11 1:31 PM >>> >>Er, he's using sipxbridge but the provider sits in his network without nat I >>think. The trace shows the provider is sending from a public ip but his pbx and all his phones have a private ip subnet. Looks li

Re: [sipx-users] Polycom 3.2.4 transfer issue

Just as an fyi, we're working a tromboning (hairpin) issue with sipXbridge... not sure if this is related or not but it is around hold / transfer / park (which are really all flavors of the same thing). Not sure when it was introduced or what is causing it at this point. Mike On Thu, Jan 27, 20

Re: [sipx-users] Polycom 3.2.4 transfer issue

On 1/27/2011 12:27 PM, Matt White wrote: >>> "Matthew Kitchin (public/usenet)" 01/27/11 12:34 PM >>> >>I think I see where the problem occurs in the sip trace, but I don't >>have the sip knowledge to understand why. >> >>Any help would be greatly appreciated. Ok, I've reviewed your traces and

Re: [sipx-users] Polycom 3.2.4 transfer issue

>>> Tony Graziano 01/27/11 1:31 PM >>> >>Er, he's using sipxbridge but the provider sits in his network without nat I >>think. The trace shows the provider is sending from a public ip but his pbx and all his phones have a private ip subnet. Looks like NAT to me. Matthew can you clarify that?

Re: [sipx-users] Polycom 3.2.4 transfer issue

>>> "Matthew Kitchin (public/usenet)" 01/27/11 >>> 12:34 PM >>> >>>Any help would be greatly appreciated. One more thing...did you change the default ports for the sipxbridge and proxy? ___ sipx-users mailing list sipx-users@list.sipfoundry.org List

Re: [sipx-users] Polycom 3.2.4 transfer issue

Er, he's using sipxbridge but the provider sits in his network without nat I think. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 4

Re: [sipx-users] Polycom 3.2.4 transfer issue

>>> "Matthew Kitchin (public/usenet)" 01/27/11 >>> 12:34 PM >>> >>I think I see where the problem occurs in the sip trace, but I don't >>have the sip knowledge to understand why. >> >>Any help would be greatly appreciated. Ok, I've reviewed your traces and your wiresharks. You've got some ser

Re: [sipx-users] Upgrading Polycom Firmware - possible FTP server missing?

Hi Tony, Thanks, strange though used to work before with 4.2.1, and I've made no config changes to the phone or DHCP. But yes changing to TFTP has solved the problem. Kind Regards, Ben Goodfellow Proprietor Tel:+44 (0)117 205 Fax:+44 (0)844 9107707 Email: b...@btg-computers.co.u

Re: [sipx-users] Upgrading Polycom Firmware - possible FTP server missing?

Um... If you change the ftp to tftp it should work. If you use ftp you should code the ip address in. ftp parameters are not pushed out in dhcp, just tftp/boot server. Or put the ftp address in. Make sure you activate the firmware. There are additional vsftpd.conf settings needed if the phone is

[sipx-users] Upgrading Polycom Firmware - possible FTP server missing?

[cid:imagec7c4f7.GIF@0dfbbedd.459ed4d2] Hi, I used to run 4.2.1 and could change my Polycom firmware by simply change which device file was active. I’ve since installed CentOS 5.5 from scratch with no options and 4.4.0, all seems to work well. Today I tried to upgrade the Polycom to 3.2.4, u

Re: [sipx-users] Calling delays on secondary server in HA cluster

From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Irena Dolovčak [irena.dolov...@gmail.com] I have a two server HA sipx cluster. When I take down the primary server, everything still works with the secondar

Re: [sipx-users] ERR in sipregistrar log

From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Peter van der Salm [peter.vanders...@smart-future.nl] We have some hard to research registration problems between SipX 4.2.1 and Unidata WPU-7700. With 4.0

Re: [sipx-users] interconnect with CS1000

I found an issue http://track.sipfoundry.org/browse/XX-6045, that looks to deal with exactly my problem. And it is marked fixed. In 4.2. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Beha

Re: [sipx-users] interconnect with CS1000

The description of the "Strip User Parameters" check box in the registrar configuration is Remove all semicolon-delimited parameters from the userpart of local SIP addresses. For example, convert 'sip:user;phone-context=x...@example.com' to 'sip:u...@example.com' when the local domain is 'exam

Re: [sipx-users] ERR in sipregistrar log

Hi Tony, Let me checkI don't see even a setting that could have to do with gruu in the phone. So I assume gruu is not supported on the Unidata WPU-7700. Peter van der Salm Smart Future cv, Buys Ballotstraat 14, 3572 ZR Utrecht, The Netherlands phone: +31 302 720 278 fax: +31 847 156 296 mo

Re: [sipx-users] Audiocodes call handling

As far as I know, it is possible if you manage mp118 manually via web interface (not via sipxconfig plugin). Try to create a huntgroup on mp118. With "channel select mode", say, ascending. Then in the "endpoint phone number table" you should mark four of your channels to belong to this huntgroup. T

Re: [sipx-users] ERR in sipregistrar log

Can you indicate if gruu is disabled on the phone or not? Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Cont

[sipx-users] ERR in sipregistrar log

Hi Folks, We have some hard to research registration problems between SipX 4.2.1 and Unidata WPU-7700. With 4.0.4 this did not happen, so apparently 'something' changed. See the transcript of an REGISTER request below. My eye was caught by Local Host:192.168.4.10 Port: -1 Port -1---

Re: [sipx-users] interconnect with CS1000

Ok, i looked at xx-7695, and i don't think it is related ... _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Thursday, January 27, 2011 3:38 PM To: Discussion list for users of sipXecs software Subject: Re: [

[sipx-users] Audiocodes call handling

Hello, I am using an Audiocodes MP 118 FXO gateway with firmware 5.60A which is working well with sipXecs 4.2. I have a situation where I need to route all outbound (from sipX to PSTN) calls through only four of the eight gateway line ports, while keeping all ports available to receive incomin

Re: [sipx-users] interconnect with CS1000

Did you look at: http://track.sipfoundry.org/browse/XX-7695 On Thu, Jan 27, 2011 at 7:23 AM, Nikolay Kondratyev wrote: > Hi all, > > i have a problem interconnecting sipx (4.2.1 and 4.4.0) to cs1000 (version > 7.00.20). > The problem is that CS1000

[sipx-users] interconnect with CS1000

Hi all, i have a problem interconnecting sipx (4.2.1 and 4.4.0) to cs1000 (version 7.00.20). The problem is that CS1000 sends invite in the following form: INVITE sip:4002;phone-context=cdp.udp@"mydomain":5080;maddr=;transport=tcp;user=phone;x-nt-redirect=redirect-server SIP/2.0 and finally sip

Re: [sipx-users] connecting to a legacy pbx

On 01/27/2011 10:14 AM, Todd Hodgen wrote: > Curious, in your implementation, are you providing call paths from Siemen > switch to Siemen switch via the Sipxecs, and is it acting as the core for > that network, or are using multiple circuits to connect all of these > switches in a different archite

Re: [sipx-users] SIPx 4.2 as an external voicemail server

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <56373> Message-ID: I had the same issue when I first setup up an external voicemail server. In my case the calls were originating

Re: [sipx-users] connecting to a legacy pbx

Curious, in your implementation, are you providing call paths from Siemen switch to Siemen switch via the Sipxecs, and is it acting as the core for that network, or are using multiple circuits to connect all of these switches in a different architecture? I think it would be a good practice for dev

Re: [sipx-users] connecting to a legacy pbx

Hi, Actually I have 2 Siemens Hipath and 3 Siemens Hicom pbxs connected to SipXecs. On each pbx I've used an E1 port connected to a Patton Smartnode 4960, with Q.SIG license. Patton gateways are configured on SipXecs as unmanaged gateways. HTH, Massimo -- Massimo Vignone UniMORE - Servi